fix(audio): fix DMC loop byte skip, add DC blocker, lazy cpal stream

Three audio bugs fixed:

1. DMC loop mode skipped the last byte of each sample iteration.
   provide_dmc_dma_byte() was immediately setting dmc_dma_request on
   loop restart while the sample buffer was still full, causing the
   while-loop in clock_cpu_cycles to service a second DMA immediately
   and overwrite the valid buffer. Per NES hardware spec, the reader
   only fills an empty buffer — the request is now left to clock_dmc
   when the output unit actually empties the buffer into the shift
   register. Fixes intermittent clicking/crackling in games that use
   looped DMC samples (BGM, SFX).

2. Missing DC blocker (high-pass filter) in AudioMixer. The NES APU
   has a capacitor-coupled output stage that blocks DC bias. Without
   it, abrupt channel state changes (length counter expiry, sweep
   mute, triangle period < 2) produce DC steps that manifest as
   audible clicks. Added a one-pole IIR high-pass filter at ~5 Hz
   applied after the existing low-pass filter.

3. cpal stream was opened at application startup with
   BufferSize::Fixed(256), forcing PipeWire/PulseAudio to run the
   entire audio graph at a 5.3 ms quantum. This disrupted other audio
   applications (browsers, media players) even when no ROM was loaded.
   Fixed by: (a) creating the stream lazily on the first push_samples
   call so no device is touched until a ROM is running, and (b)
   switching to BufferSize::Default so the audio server chooses the
   quantum instead of the emulator imposing one. Ring buffer capacity
   increased from 1536 to 4096 samples to absorb larger server quanta.

crates/nesemu-desktop/src/main.rs

@@ -20,7 +20,7 @@ const APP_ID: &str = "org.nesemu.desktop";
 const TITLE: &str = "NES Emulator";
 const SCALE: i32 = 3;
 const SAMPLE_RATE: u32 = 48_000;
-const AUDIO_RING_CAPACITY: usize = 1536;
+const AUDIO_RING_CAPACITY: usize = 4096;
 const AUDIO_CALLBACK_FRAMES: u32 = 256;
 
 fn main() {
@@ -482,16 +482,26 @@ struct CpalAudioSink {
 impl CpalAudioSink {
     fn new(volume: Arc<AtomicU32>) -> Self {
         let ring = Arc::new(RingBuffer::new(AUDIO_RING_CAPACITY));
-        let ring_for_cb = Arc::clone(&ring);
-        let vol_for_cb = Arc::clone(&volume);
-        let stream = Self::try_build_stream(ring_for_cb, vol_for_cb);
+        // Do NOT open the audio device here. Creating a cpal stream at startup
+        // forces the system audio server (PipeWire/PulseAudio) to allocate
+        // resources and may disrupt other running audio applications even when
+        // the emulator is idle. The stream is opened lazily on the first
+        // push_samples call, i.e. only when a ROM is actually playing.
         Self {
-            _stream: stream,
+            _stream: None,
             ring,
             _volume: volume,
         }
     }
 
+    fn ensure_stream(&mut self) {
+        if self._stream.is_none() {
+            let ring_for_cb = Arc::clone(&self.ring);
+            let vol_for_cb = Arc::clone(&self._volume);
+            self._stream = Self::try_build_stream(ring_for_cb, vol_for_cb);
+        }
+    }
+
     fn try_build_stream(ring: Arc<RingBuffer>, volume: Arc<AtomicU32>) -> Option<cpal::Stream> {
         use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
 
@@ -548,6 +558,7 @@ impl CpalAudioSink {
 
 impl nesemu::AudioOutput for CpalAudioSink {
     fn push_samples(&mut self, samples: &[f32]) {
+        self.ensure_stream();
         self.ring.push(samples);
     }
 }
@@ -567,7 +578,10 @@ fn cpal_stream_config() -> cpal::StreamConfig {
     cpal::StreamConfig {
         channels: 1,
         sample_rate: cpal::SampleRate(SAMPLE_RATE),
-        buffer_size: cpal::BufferSize::Fixed(AUDIO_CALLBACK_FRAMES),
+        // Use the audio server's default buffer size to avoid forcing the entire
+        // PipeWire/PulseAudio graph into low-latency mode, which would disturb
+        // other audio applications (browsers, media players, etc.).
+        buffer_size: cpal::BufferSize::Default,
     }
 }
 
@@ -803,9 +817,9 @@ mod tests {
     }
 
     #[test]
-    fn desktop_audio_ring_budget_stays_below_25ms() {
+    fn desktop_audio_ring_budget_stays_below_100ms() {
         let latency_ms = audio_ring_latency_ms(AUDIO_RING_CAPACITY, SAMPLE_RATE);
-        let max_budget_ms = 40.0;
+        let max_budget_ms = 100.0;
         assert!(
             latency_ms <= max_budget_ms,
             "desktop audio ring latency budget too high: {latency_ms:.2}ms"
@@ -813,12 +827,11 @@ mod tests {
     }
 
     #[test]
-    fn desktop_audio_uses_fixed_low_latency_callback_size() {
+    fn desktop_audio_uses_default_buffer_size() {
         let config = cpal_stream_config();
-        assert_eq!(
-            config.buffer_size,
-            cpal::BufferSize::Fixed(AUDIO_CALLBACK_FRAMES)
-        );
+        // Default lets the audio server (PipeWire/PulseAudio) choose the
+        // buffer size, preventing interference with other audio applications.
+        assert_eq!(config.buffer_size, cpal::BufferSize::Default);
     }
 
     #[test]

src/native_core/apu/api.rs

@@ -200,9 +200,15 @@ impl Apu {
         }
         if self.dmc_bytes_remaining == 0 {
             if (self.io[0x10] & 0x40) != 0 {
+                // Loop mode: reset address and byte counter.
+                // Do NOT request another DMA here — the sample buffer is full
+                // right now. clock_dmc will request the next fetch when the
+                // output unit empties the buffer into the shift register, which
+                // is the correct NES hardware behaviour (reader only fills an
+                // empty buffer).  Requesting early would overwrite the valid
+                // buffer and skip the last byte of each loop iteration.
                 self.dmc_bytes_remaining = self.dmc_sample_length_bytes();
                 self.dmc_current_addr = self.dmc_sample_start_addr();
-                self.dmc_dma_request = true;
             } else if self.dmc_irq_enabled {
                 self.dmc_irq_pending = true;
             }

src/runtime/audio.rs

@@ -11,12 +11,21 @@ pub struct AudioMixer {
     // Coefficient: a = exp(-2π * fc / fs). At fc=14000, fs=48000: a ≈ 0.160
     lp_coeff: f32,
     lp_state: f32,
+    // One-pole IIR high-pass filter (DC blocker). Removes the DC bias that
+    // accumulates when APU channels switch state, preventing audible clicks and
+    // pops. Approximates the NES capacitor-coupled output stage (~5 Hz cutoff).
+    // Formula: y[n] = hp_coeff * y[n-1] + x[n] - x[n-1]
+    // Coefficient: a = exp(-2π * fc / fs). At fc=5, fs=48000: a ≈ 0.99935.
+    hp_coeff: f32,
+    hp_prev_x: f32,
+    hp_prev_y: f32,
 }
 
 impl AudioMixer {
     pub fn new(sample_rate: u32, mode: VideoMode) -> Self {
         let cpu_hz = mode.cpu_hz();
         let lp_coeff = (-2.0 * std::f64::consts::PI * 14_000.0 / sample_rate as f64).exp() as f32;
+        let hp_coeff = (-2.0 * std::f64::consts::PI * 5.0 / sample_rate as f64).exp() as f32;
         Self {
             sample_rate,
             samples_per_cpu_cycle: sample_rate as f64 / cpu_hz,
@@ -24,6 +33,9 @@ impl AudioMixer {
             last_output_sample: 0.0,
             lp_coeff,
             lp_state: 0.0,
+            hp_coeff,
+            hp_prev_x: 0.0,
+            hp_prev_y: 0.0,
         }
     }
 
@@ -35,6 +47,8 @@ impl AudioMixer {
         self.sample_accumulator = 0.0;
         self.last_output_sample = 0.0;
         self.lp_state = 0.0;
+        self.hp_prev_x = 0.0;
+        self.hp_prev_y = 0.0;
     }
 
     pub fn push_cycles(&mut self, cpu_cycles: u32, channels: ChannelOutputs, out: &mut Vec<f32>) {
@@ -56,17 +70,23 @@ impl AudioMixer {
         let a = self.lp_coeff;
         let b = 1.0 - a;
         if samples == 1 {
-            let s = a * self.lp_state + b * sample;
-            self.lp_state = s;
-            out.push(s);
+            let lp = a * self.lp_state + b * sample;
+            self.lp_state = lp;
+            let hp = self.hp_coeff * self.hp_prev_y + lp - self.hp_prev_x;
+            self.hp_prev_x = lp;
+            self.hp_prev_y = hp;
+            out.push(hp);
         } else {
             let denom = samples as f32;
             for idx in 0..samples {
                 let t = (idx + 1) as f32 / denom;
                 let interp = start + (sample - start) * t;
-                let s = a * self.lp_state + b * interp;
-                self.lp_state = s;
-                out.push(s);
+                let lp = a * self.lp_state + b * interp;
+                self.lp_state = lp;
+                let hp = self.hp_coeff * self.hp_prev_y + lp - self.hp_prev_x;
+                self.hp_prev_x = lp;
+                self.hp_prev_y = hp;
+                out.push(hp);
             }
         }
         self.last_output_sample = sample;